Systems and methods for adaptive beamforming

ABSTRACT

A system for capturing a user&#39;s voice, including a plurality of microphones disposed about a vehicle cabin, each of the plurality of microphones generating a microphone signal, wherein the vehicle cabin defines a plurality of seating positions; a voice-activity detector configured to detect when, at least, a user seated in a target seat of the plurality of seating positions is speaking; and an adaptive beamformer configured to receive the microphone signals from the plurality of microphones and to generate, based on the microphone signals and a noise coherence matrix, an estimate of the acoustic signal at the target seating position, according to an adaptive beamforming algorithm, wherein the noise coherence matrix ceases to be updated when, according to the voice-activity detector, the user seated in the target seating position is speaking.

BACKGROUND

The present disclosure generally relates to systems and methods foradaptive beamforming.

SUMMARY

All examples and features mentioned below can be combined in anytechnically possible way.

According to an aspect, an adaptive beamforming system includes: aplurality of microphones disposed about a vehicle cabin, each of theplurality of microphones generating a microphone signal, wherein thevehicle cabin defines a plurality of seating positions; a voice-activitydetector configured to detect when, at least, a user seated in a targetseat of the plurality of seating positions is speaking; and an adaptivebeamformer configured to receive the microphone signals from theplurality of microphones and to generate, based on the microphonesignals and a noise coherence matrix, an estimate of an acoustic signalat the target seating position, according to an adaptive beamformingalgorithm, wherein the noise coherence matrix ceases to be updated when,according to the voice-activity detector, the user seated in the targetseating position is speaking.

In an example, the target seat is selected according to a userselection.

In an example, the target seat is selected according to which seatingposition of the plurality of seating positions the voice activitydetector detects a user speaking.

In an example, the adaptive beamformer is further configured to ceaseupdating the noise coherence matrix when, according to thevoice-activity detector, a user in any of the plurality of seatingpositions is speaking.

In an example, the adaptive beamformer is further configured tocalculate a plurality of noise coherence matrices, each of the pluralityof noise coherence matrices being representative of a noise condition ata respective associated seating position of the plurality of seatingpositions, wherein each of the plurality of noise coherence matricesceases to be updated when, according to the voice-activity detector, auser seated in the associated seating position is speaking.

In an example, the adaptive beamformer is further configured to generatean estimate of a second acoustic signal at a second target seatingposition, when, according to the voice-activity detector, the userseated in the target seating position is speaking and the user seated inthe second target seating position is speaking, wherein the estimate ofthe acoustic signal and the second acoustic signal being summedtogether.

In an example, the adaptive beamformer is further configured to generatean estimate of the acoustic signal and a second acoustic signal at asecond target seating position, according to a second adaptivebeamforming algorithm, when, according to the voice-activity detector,the user seated in the target seating position is speaking and the userseating in the second target seating position is speaking.

In an example, the second adaptive beamforming algorithm is a linearlyconstrained minimum variance beamforming algorithm.

According to another aspect, an adaptive beamforming system includes: aplurality of microphones disposed about a vehicle cabin, each of theplurality of microphones generating a microphone signal, wherein thevehicle cabin defines a plurality of seating positions; a voice-activitydetector configured to detect when a user seated in a target seatingposition of the plurality of seating positions is speaking; and anadaptive beamformer configured to receive the microphone signals fromthe plurality of microphones and to generate, based on the microphonesignals and one of a first noise coherence matrix and a second noisecoherence matrix, an estimate of an acoustic signal at the targetseating position, according to an adaptive beamforming algorithm,wherein the first noise coherence matrix is computed by recursivelysumming a first previously-calculated noise coherence matrix with afirst newly-calculated noise coherence matrix, wherein the firstpreviously-calculated noise coherence matrix is weighted more heavilythan the first newly-calculated noise coherence matrix, whereincoefficients of the adaptive beamformer are updated using the firstnoise coherence matrix when the voice-activity detector detects that theuser seated in the target seating position is speaking, wherein thesecond noise coherence matrix is computed by recursively summing asecond previously-calculated noise coherence matrix with a secondnewly-calculated noise coherence matrix, wherein the secondpreviously-calculated noise coherence matrix is weighted less heavilythan the second newly-calculated noise coherence matrix, wherein thecoefficients of the adaptive beamforming filter are updated using thesecond noise coherence matrix when the voice-activity detector does notdetect that the user seated in the target seating position is speaking.

In an example, the coefficients of the adaptive beamformer are updatedusing a historic first noise coherence matrix stored during a previousframe, wherein the historic first noise coherence matrix was stored atleast a predetermined period of time before the voice-activity detectordetected that the user seated in the target seating position isspeaking, wherein the predetermined period of time is greater than adelay required for the voice-activity detector to detect that the targetuser is speaking.

According to another aspect, an adaptive beamforming system includes: aplurality of microphones disposed about a vehicle cabin, each of theplurality of microphones generating a microphone signal, wherein thevehicle cabin defines a plurality of seating positions; and an adaptivebeamformer configured to receive the microphone signals from theplurality of microphones and to generate, based on the microphonesignals and a noise coherence matrix, an estimate of an acoustic signalat a target seating position, according to an adaptive beamformingalgorithm, wherein the noise coherence matrix is determined, at least inpart, from a predetermined noise coherence matrix, the predeterminednoise coherence matrix being representative of a noise conditionintroduced to the vehicle cabin by at least one speaker disposed in thevehicle cabin.

In an example, a gain of the predetermined noise coherence matrix is setaccording to a magnitude of the noise condition.

In an example, the gain of the predetermined noise coherence matrix isincreased over a plurality of samples.

In an example, the predetermined noise coherence matrix is subtractedfrom an updated noise coherence matrix when the noise condition ceasesto be introduced to the vehicle cabin, wherein the updated noisecoherence matrix is an update to the noise coherence matrix according tothe plurality of microphone signals.

In an example, the noise coherence matrix comprises a sum of acalculated noise coherence matrix and the predetermined noise coherencematrix, wherein the calculated noise coherence matrix is inverted froman inverted state before being summed with the predetermined noisecoherence matrix.

In an example, the predetermined noise coherence matrix is retrieved asan inverted matrix.

According to another aspect, an adaptive beamforming system includes: aplurality of microphones disposed about a vehicle cabin, each of theplurality of microphones generating a microphone signal, wherein thevehicle cabin defines a plurality of seating positions; an adaptivebeamformer configured to receive the microphone signals from theplurality of microphones and to generate, based on the microphonesignals and a noise coherence matrix, an estimate of an acoustic signalat a target seating position, according to an adaptive beamformingalgorithm, wherein an artificial white noise signal is added to each ofthe plurality of microphone signals such that white noise gain of theestimate of the acoustic signal is improved.

In an example, the artificial white noise signal is selected such that aminimum white noise gain is achieved across a predetermined frequencyrange.

In an example, the artificial white noise signal is predetermined.

In an example, the artificial white noise signal is selected from aplurality of artificial white noise signals according to the noisecondition within the cabin.

In an example, the artificial white noise signal is adjusted such that acondition number of the noise coherence matrix is maintained within apredetermined range across frequency.

The details of one or more implementations are set forth in theaccompanying drawings and the description below. Other features,objects, and advantages will be apparent from the description and thedrawings, and from the claims.

BRIEF DESCRIPTION OF THE DRAWINGS

In the drawings, like reference characters generally refer to the sameparts throughout the different views. Also, the drawings are notnecessarily to scale, emphasis instead generally being placed uponillustrating the principles of the various aspects.

FIG. 1A depicts a schematic drawing of a vehicle cabin with a controllerimplementing an adaptive beamformer and microphones, according to anexample.

FIG. 1B depicts a schematic drawing of a controller implementing anadaptive beamformer and voice activity detector and receiving microphonesignals from microphones, according to an example.

FIG. 2A depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2B depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2C depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2D depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2E depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2F depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 2G depicts a method for adaptive beamforming in which a noisecoherence matrix is updated when a voice activity detector detects thatat least one user is not speaking, according to an example.

FIG. 3A depicts a method for adaptive beamforming in which noisecoherence matrices with variable forgetting factors are selectedaccording to the voice activity detector of a least one user, accordingto an example.

FIG. 3B depicts a method for adaptive beamforming in which noisecoherence matrices with variable forgetting factors are selectedaccording to the voice activity detector of a least one user, accordingto an example.

FIG. 3C depicts a method for adaptive beamforming in which noisecoherence matrices with variable forgetting factors are selectedaccording to the voice activity detector of a least one user, accordingto an example.

FIG. 3D depicts a method for adaptive beamforming in which noisecoherence matrices with variable forgetting factors are selectedaccording to the voice activity detector of a least one user, accordingto an example.

FIG. 4A depicts a method for adaptive beamforming in which the noisecoherence matrix is, at least in part, based on a predetermined noisecoherence matrix, according to an example.

FIG. 4B depicts a method for adaptive beamforming in which the noisecoherence matrix is, at least in part, based on a predetermined noisecoherence matrix, according to an example.

FIG. 5 depicts a method for adaptive beamforming in which artificialwhite noise is added to the microphone signals, according to an example.

DETAILED DESCRIPTION

A microphone array constitutes one of the first stages in the signalprocessing chain of a hands-free system disposed in a vehicle. Inaddition to their noise and interference rejection capability,microphone arrays lead to an improved performance of the subsequentcomponents in the system such as the acoustic echo canceller and theresidual echo suppressor due to the improved signal-to-noise ratios attheir inputs. The vehicle environment is very noisy, and the use of amicrophone array is thus essential to obtain an acceptable overallsystem performance.

Microphone arrays can be either fixed or adaptive. Fixed microphonearrays are usually easier to implement and are less computationallyexpensive. However, they suffer from inconsistent performance whendealing with varying noise conditions. Indeed, it is extremely difficultto design a fixed microphone that can perform well in all noiseconditions. The array has the best performance when it is applied to thesame noise condition it was designed for. Once the noise conditionchanges, however, the performance of the array starts deterioratinguntil it reaches a point where the array is hurting the performanceinstead of helping it.

An adaptive beamformer removes the performance degradation of a fixedbeamformer by adapting the filter coefficients to the noise condition.FIG. 1A depicts an example schematic diagram of a vehicle cabin 100featuring a controller 102 that receives multiple microphone signals m₁,m₂, m₃, from an array of microphones 104 and implements an adaptivebeamforming algorithm to estimate an acoustic signal at at least one ofthe seating positions. The cabin defines a plurality of seatingpositions, shown in FIG. 1A as P1 and P2 respectively centered on seats106, however any number of seating positions can be so defined (e.g.,five seating positions can be defined for a five-seat passengervehicle). Similarly, although the array of microphones 104 is shown toinclude only three microphones 104, the array can include any number ofmicrophones, disposed in any location suitable for capturing speechactivity within the cabin, and producing any number of microphonesignals m. For the purposes of this disclosure, a microphone is anysensor receiving an acoustic signal within the vehicle cabin andtransducing it into an electrical signal.

In an example, controller 102 can comprise a non-transitory storagemedium 108 and processor 110. In an example, non-transitory storagemedium 108 can store program code that, when executed by processor 110,implements the various filters and algorithms described below.Controller 102 can be implemented in hardware and/or software. Forexample, controller 102 can be implemented by a SHARC floating-point DSPprocessor, but it should be understood that controller 102 can beimplemented by any other processor, FPGA, ASIC, or other suitablehardware.

FIG. 1B depicts a block diagram of the controller 102 and microphones104. As shown, controller 102 can implement an adaptive beamformer 112,according to an adaptive beamforming algorithm, that is configured toreceive the microphone signals m and to produce an output signal p_(out)representative of an estimate of the acoustic signal at at least oneseating position. For the purposes of this disclosure, adaptivebeamformer 112 estimates at least one acoustic signal by filtering eachmicrophone signal m to account for the phase-shift and attenuation ofthe acoustic signal at the location of the respective microphone 104. Inaddition, adaptive beamformer 112, according to the adaptive beamformeralgorithm, reduces the performance degradation of a fixed beamformer byadapting the filter coefficients to the noise condition within thecabin, such that ambient noise is reduced in the output signal p_(out).Thus, the estimate of the acoustic signal is not a pure estimate, butrather one that reduces ambient noise in order to clearly reproduce auser's voice. The operation of adaptive beamformer 112 will be morefully described in connection with FIGS. 2-5, as detailed below.

As shown in FIG. 1B controller 102 can further implement voice activitydetector 114. Voice activity detector 114, in one example, can receivesome or all of microphone signals m and, from them, determine whetherthe user seated in seating position P1 is speaking (voice activity).Voice activity detector 114 can output a voice activity detection signalv_(out) representing whether the user is speaking (e.g., VAD=0 orVAD=1). In addition, in some examples, voice activity detector candetect voice activity in other seating positions (e.g., seating positionP2) in the vehicle. In these examples, voice activity detection signalv_(out) can identify which seating position the voice activity isdetected. Such a per-seat voice activity detector can detect any numberof voice activities within the vehicle cabin 100, i.e., VAD_(i), i=1, .. . , N_(s), where VAD_(i) detects speech from the ith desired seat andN_(s) is the number of desired seats.

Voice activity detection methods are generally known, and voice activitydetector 114 can be implemented according to any suitable voice activitydetection method. Although the voice activity detector 114 is shownimplemented by controller 102, in various alternative examples, thevoice activity detector can be implemented separate from controller 102.

FIGS. 2-5 depict methods featuring various improvements to adaptivebeamforming. These methods are described in connection with the systemof FIGS. 1A and 1B; however it should be understood that the steps ofthe methods of FIGS. 2-5 can be implemented by any suitable system foradaptive beamforming.

Turning first to FIG. 2A there is shown a method 200 for adaptivebeamforming in which a noise coherence matrix is only updated when auser in a target seat is not speaking. At step 202, a plurality ofmicrophone signals m are received from microphones 104 disposed aboutthe vehicle cabin 100. At step 204, adaptive beamformer 112 employsthese microphone signals to generate, based on a noise coherence matrix,an estimate of an acoustic signal at the target seating positionaccording to an adaptive beamforming algorithm. (It should be understoodthat the “target seating position” can refer to any seating positionwithin the vehicle.)

For example, when the user at seating position P₁ is speaking, eachmicrophone 104 will receive an acoustic signal resulting from the userspeech that is affected by the path from the user at seating position P₁and the respective microphone 104. The nature of this path (e.g.,including the length of the path and any obstructions positioned withinthe path) will dictate some change in the relative magnitude and phaseof each microphone signal m. As shown in FIG. 1A, microphone 104 a,producing microphone signal m₁, is disposed at a location some distanced₁ from seating position P₁. Likewise, microphone 104 b is disposed at alocation some distance d₂ from seating P₁, and microphone 104 c isdisposed at a location some distance d₃ from seating position P₁.Assuming that the path from the microphone 104 a and 104 c to seatingposition P₁ is clear of any obstructions, because distance d₁ is shorterthan distance d₃, microphone 104 a will experience attenuation and phaseshift of the acoustic signal that is relatively smaller than theattenuation and phase shift of the same acoustic signal received atmicrophone 104 b. Adaptive beamformer 112, in effect, filters eachmicrophone signal m to remove the attenuation and phase shift resultingfrom the path from seating position P₁ to the respective microphone. Inthis way, microphone signal m₃ is filtered to remove the attenuation andphase shift of the acoustic signal at microphone 104 c as a result ofthe acoustic signal traveling distance d₃, microphone signal m₂ isfiltered to remove the attenuation and phase shift of the acousticsignal at microphone 104 b as a result of the acoustic signal travelingdistance d₂, and microphone signal m₁ is filtered to remove theattenuation and phase shift of the acoustic signal at microphone 104 aas a result of the acoustic signal traveling distance d₁. To the extentthat the path between seating position P₁ is not clear of obstructions,the attenuation and phase shift induced by the obstruction can also beestimated and accounted for in the filter.

Alternatively, one microphone, such as microphone 104 a, can serve as areference microphone for seating position P₁ and every other microphonecan be filtered to account to the relative attenuation and phase shiftto microphone signal m₁. Thus, microphone m₂ can be filtered to removethe attenuation and phase shift with respect to microphone signal m₁ andmicrophone m₃ can be filtered to remove the attenuation and phase shiftwith respect to microphone signal m₁. In this case, adaptive beamformer112 estimates the magnitude and phase relationship between eachmicrophone and filters each microphone signal m to constructively alignthe microphone signal m with a reference microphone signal in magnitudeand phase.

In either example—i.e., accounting for the unique transfer function fromeach microphone and the seating position or accounting for the magnitudeand phase relationship between each microphone and a referencemicrophone—the adaptive beamformer 112 is considered to steer a beamtoward seating position P₁ to estimate the acoustic signal of the userseated there. Further, each microphone signal must be filtered uniquelyfor each frequency, as the attenuation and phase shift change acrossfrequency. In addition, adaptive beamformer 112 can account forreflections within the vehicle cabin resulting from the user speaking atseating position P₁. For example, speech from a user seated at seatingposition P₁ will generate reflections from the windows, ceiling, etc. Toaccount for these reflections, adaptive beamformer 112 can steeradditional beams toward the source of the reflections in the same mannerthat a beam was steered toward the seating position. Adaptive beamformer112 can thus account for the accumulation of all sources of the desiredspeech signal. (The above example is described for a target seat P₁,however, any seat within the vehicle can similarly serve as the targetseat at which the acoustic signal is estimated.)

As described above, fixed beamformer performance and the quality ofestimated speech will degrade in the presence of noise within thevehicle cabin. Adaptive beamforming algorithms reduce the performancedegradation of a fixed beamformer by adapting the filter coefficients tothe noise condition within the cabin. One such adaptive beamformeralgorithm is a minimum variance distortionless response (MVDR). The MVDRdesign coefficients at the kth frequency bin can be obtained as follows:

$\begin{matrix}{{{w^{H}(k)} = \frac{{C_{nn}^{- 1}(k)} \cdot {d(k)}}{{d^{H}(k)} \cdot {C_{nn}^{- 1}(k)} \cdot {d(k)}}},} & (1)\end{matrix}$

where C_(nn)(k) is an N×N noise coherence matrix at frequency k, with Nbeing the number of microphones, and d(k) is the steering vector of thedesired source at the same frequency. As described above, the steeringvector is the representation of the delays and the attenuations whichdepends on the locations of the microphones and the desired source. Thenoise coherence matrix represents the detected noise condition for thecurrent sample.

In practice, d(k) and the initial C_(nn)(k) can be generated fromrecordings of the desired and undesired sources in the vehicle. Usingthe recordings instead of the free-field equations results in a morerobust design since the recordings incorporate the reflections andtransfer functions in the vehicle. Steering vector d(k) is measured apriori using recordings of the desired signal and stored in the memory.This is possible since the desired source location (e.g. driver seat)does not undergo a large change over time. The noise coherence matrices,by contrast, are updated regularly to track the noise condition. Thenoise coherence update equation is expressed as follows:

C _(nn)(k,n)=λ(k)·C _(nn)(k,n−1)+[1−λ(k)]·x(k,n)·x ^(H)(k,n),  (2)

where λ(k) is a forgetting factor which controls the speed ofadaptation, C_(nn)(k,n) is the updated noise coherence matrix at the kthfrequency bin, C_(nn)(k,n−1) is the previous noise coherence matrix, andx(k,n) is the received signal vector at the current frame. The noisecoherence matrix can thus be updated recursively using the previouslycalculated noise coherence matrix and a newly calculated noise coherencematrix.

The forgetting factor λ(k) controls the speed of convergence at thefrequency bin k. λ(k) provides a tradeoff between the speed ofconvergence to the correct noise condition and the long-term performanceof the algorithm, especially in the presence of speech activity.

The matrix inversion of the noise coherence matrix (i.e., C_(nn) ⁻¹(k))of Eq. (1) has to be done for each frequency. For a large number ofmicrophones 104 or for a large number of FFT points, this results in ahigh computational complexity. Using the matrix inversion lemma, a moreefficient implementation of the matrix inversion can be implemented.Accordingly, the inverse of the noise coherence can be computed asfollows

C _(nn) ⁻¹(k,n)=λ(k)⁻¹ ·C _(nn) ⁻¹(k,n−1)−ΔC _(nn) ⁻¹(k,n)  (3)

where C_(nn) ⁻¹(k,n−1) is the inverse of the noise coherence matrix atthe previous frame and ΔC_(nn) ⁻¹(k,n) is expressed as follows:

$\begin{matrix}{{\Delta\;{C_{nn}^{- 1}( {k,n} )}} = {\frac{\begin{matrix}{{\lambda(k)}^{- 2} \cdot {C_{nn}^{- 1}( {k,{n - 1}} )} \cdot \lbrack {1 - {\lambda(k)}} \rbrack \cdot} \\{x{( {k,n} ) \cdot {x^{H}( {k,n} )} \cdot {C_{nn}^{- 1}( {k,{n - 1}} )}}}\end{matrix}}{1 + {{\lambda(k)}^{- 1} \cdot \lbrack {1 - {\lambda(k)}} \rbrack \cdot {x^{H}( {k,n} )} \cdot {C_{nn}^{-}( {k,{n - 1}} )} \cdot {x( {k,n} )}}}.}} & (4)\end{matrix}$

Eqs. (3) and (4) can thus be used to recursively calculate the inverseof the noise coherence matrix in a more efficient manner.

The noise coherence matrix in Eqs. (1)-(2) can be initialized to a knownnoise coherence matrix, i.e. C_(nn)(k,0)=C_(nn)(k,init), k=1 . . .N_(f), where N_(f) is the number of frequency points and C_(nn)(k,init)is a noise coherence matrix that is computed a priori (i.e.,predetermined) for the expected noise condition in the vehicle. In asimilar fashion, if the matrix inversion lemma is used to update theinverse of the noise coherence matrix (Equations (3)-(4)), C_(nn)⁻¹(k,0) can be initialized to the known noise condition, i.e. C_(nn)⁻¹(k,0)=C_(nn) ⁻¹(k,init). In one example, the predetermined noisecoherence matrix and inverse noise coherence matrix can be factorydefault values that are selected to work well for most noise situations.In an alternative example, the noise coherence matrix and inverse noisecoherence matrix can be stored in a non-transitory storage medium (e.g.,non-transitory storage medium 108) at predetermined intervals or whenthe vehicle is turned off. When the noise coherence matrix and theinverse noise coherence matrix is initialized, the most recent storednoise coherence matrix and inverse noise coherence matrix is retrievedfrom storage and used. In an alternative example, only one of the noisecoherence matrix or inverse noise coherence matrix is stored andretrieved from storage and inverted to arrive at the other. Once thenoise coherence matrix and the inverse noise coherence matrix isinitialized, these values are updated, as described above, to track theactual noise condition.

However, continuing to update the noise coherence matrix in the presenceof the desired signal (i.e., voice activity at the target seat) wouldresult in the undesired nulling of the acoustic voice signal. Tocircumvent this, the noise coherence matrix can be only updated in theabsence of the desired signal as shown in the following equation:

$\begin{matrix}{{C_{nn}( {k,n} )} = \{ \begin{matrix}{{{{\lambda(k)} \cdot {C_{nn}( {k,{n - 1}} )}} + {\lbrack {1 - {\lambda(k)}} \rbrack \cdot {x( {k,n} )} \cdot {x^{H}( {k,n} )}}},} & {{VAD} = 0} \\{\mspace{371mu}{{C_{nn}( {k,{n - 1}} )},}} & {{VAD} = 1}\end{matrix} } & (5)\end{matrix}$

Stated differently, in the absence of voice activity (i.e., VAD=0), thereceived signals m from microphones 104 are a good representation of thenoise, and updating the noise coherence matrix will, accordingly, trackthe current condition of the noise, as in Eq. (2). On the other hand, inthe presence of voice activity (VAD=1), the received signal frommicrophones 104 consists of both speech and noise and, as a result, isnot a good representation of the noise. Accordingly, in the presence ofvoice activity, the noise coherence matrix from the previous sample isset as the noise coherence matrix for the present sample. Likewise, foreach successive sample in which the voice activity continues, the noisecoherence matrix from the previous sample is set as the current noisecoherence matrix. Thus, the noise coherence matrix calculated before thestart of the voice activity is used for the duration of the detectedvoice activity. Once the voice activity ceases, the noise coherencematrix is again updated according to Eq. (5).

As described above, the matrix inversion lemma can be used to calculatethe inverse of the noise coherence matrix in a more efficient manner.Using the voice activity detection to prevent updating the inverse ofthe noise coherence matrix in the presence of the desired signal usingthe matrix inversion lemma becomes:

$\begin{matrix}{{C_{nn}^{- 1}( {k,n} )} = \{ \begin{matrix}{{{{\lambda(k)}^{- 1} \cdot {C_{nn}^{- 1}( {k,{n - 1}} )}} - {\Delta\;{C_{nn}^{- 1}( {k,n} )}}},} & {{VAD} = 0} \\{\mspace{225mu}{{C_{nn}^{- 1}( {k,{n - 1}} )},}} & {{VAD} = 1}\end{matrix} } & (6)\end{matrix}$

where ΔC_(nn) ⁻¹(k,n) is calculated according to Eq. (4).

FIG. 2B depicts the steps of updating the noise-coherence matrix whenthe user in the target seat is speaking. At step 206, conditional step206 asks whether voice activity is detected at the target seatingposition. If not, the noise coherence matrix is updated according to thenoise detected at the current sample (see., e.g., Eq. (12)). Otherwise,method 200 proceeds to step 204 without updating noise coherence matrix(that is, the previously calculated noise coherence matrix is used).

The formulation above using the voice-activity detection can be extendedto cover multiple seats in the vehicle. In other words, adaptivebeamformer 112 can be configured to detect an acoustic signal at morethan one seat in the vehicle.

Generally, the steering vectors per seat, i.e. d_(i)(k), can be computeda priori and stored in the memory, as the location of the seats does notsubstantially vary. In one example, a user can have the ability toselect the target seat, and as a result, the corresponding steeringvectors are used in the equation. Thus, if the seating position P2 isselected, the steering vectors for seating position P2 are used in Eq.(1). Whereas, if seating position P₁ is selected, the steering vectorsfor seating position P1 are used in Eq. (1). This is depicted in theflowchart of FIG. 2C, which shows, at step 210, preceding step 202, auser selection of a target seat is received. This user selectiondetermines the target seat for which the acoustic signal is estimated atstep 204. The user selection can be received according to any suitableuser interface (e.g., using a touchscreen in the vehicle).

In a second example, the selected steering vectors can be chosen basedon the active speech location. In other words, when speech is detectedfrom a particular seat i (VAD_(i)=1), the corresponding steering vectorsare used. Thus, in a two-seat example, Eq. (1) is modified as follows:

$\begin{matrix}{{w^{H}(k)} = \{ \begin{matrix}\frac{{C_{nn}^{- 1}(k)} \cdot {d_{1}(k)}}{{d_{1}^{H}(k)} \cdot {C_{nn}^{- 1}(k)} \cdot {d_{1}(k)}} & {{VAD}_{1} = 1} \\\frac{{C_{nn}^{- 1}(k)} \cdot {d_{2}(k)}}{{d_{2}^{H}(k)} \cdot {C_{nn}^{- 1}(k)} \cdot {d_{2}(k)}} & {{VAD}_{2} = 1}\end{matrix} } & (7)\end{matrix}$

where d₁ is the steering vector associated with the first seatingposition P1 and d₂ is the steering vector associated with the secondseating position P2. Thus, the target seat at which the acoustic signalis estimated is determined according to which seat voice activity isdetected. Of course, this can be expanded for any number of seats—twoseats are only provided as an example. This is depicted in FIG. 2D, inwhich at step 212 the target seat (selected according to the steeringvector) is selected according to which seating position the voiceactivity detector detects a user speaking.

If speech is active in more than one seat, (for example, both VAD₁=1 andVAD₂=1) the filter coefficients can be given by the sum of equations ofEq. (7). That is, the coefficients can be given sum of w^(H)(k)calculated for each steering vector d₁, d₂ as follows:

$\begin{matrix}{{w^{H}(k)} = {\frac{{C_{nn}^{- 1}(k)} \cdot {d_{1}(k)}}{{d_{1}^{H}(k)} \cdot {C_{nn}^{- 1}(k)} \cdot {d_{1}(k)}} + \frac{{C_{nn}^{- 1}(k)} \cdot {d_{2}(k)}}{{d_{2}^{H}(k)} \cdot {C_{nn}^{- 1}(k)} \cdot {d_{2}(k)}}}} & (8)\end{matrix}$

This likewise can be extended for any number of active speakers withinthe vehicle. For example, as depicted in FIGS. 2E-2F, if, at conditionalstep 214, voice activity is detected at only one seating position (i.e.,branch NO) the target seat is selected according to which seatingposition voice activity is detected, as described above in connectionwith step 212. However, if voice activity is detected at more than oneseating position, at step 216 (FIG. 2F), an estimate of each acousticsignal at seating position at which voice activity is detected isgenerated according to the adaptive beamforming algorithm. Theseestimates can be summed together, e.g., as described in Eq. (8) above.

Alternatively, as shown in step 218 of FIG. 2G, a different beamformingalgorithm can be implemented if speech is active in more than one seat.For example, a linearly constrained minimum variance (LCMV) formulationcan be used instead of MVDR. Thus, if speech is active in only one seat,MVDR can be used according to, for example, Eq. (7) and step 212;however, if speech is active in more than one seat, LCMV (which isadapted to estimate more than one acoustic signal) can be used toestimate the speech of multiple speakers.

With regard to updating the noise coherence matrix when one of multipleseats can be selected, in one example, common noise coherence matricescan be employed across all desired seats that are updated when the voiceactivity detectors show speech is absent (i.e., VAD_(i)=0, i=1 . . .N_(s)). This method reduces the required memory and computationalcomplexity. However, it does not allow cancellation of speech at any ofthe other desired seats when a user at a selected seat position isspeaking.

Alternatively, noise coherence matrices per seat, i.e. C_(i,nn)(k,n) canbe calculated. For example, when seating position P1 is selected as thetarget seat, a noise coherence matrix C_(1,nn)(k) for the first seatingposition is used, which cancels speech at other seating positions;whereas, when seating position P2 is selected as the target seat, anoise coherence matrix C_(1,nn)(k), which cancels speech at otherseating positions is used. Each noise coherence matrix can be updatedwhen the corresponding voice activity detector shows absence of speech.Thus, for example, C_(1,nn)(k) can be updated when VAD₁=0, C_(2,nn)(k)can be updated when VAD₂=0, etc.

As described above, a smaller forgetting factor λ(k) results in fastertracking of the noise condition, but at the expense of speechdegradation during speech activity. On the other hand, a largerforgetting factor adapts slower to changes in the noise condition buthas less speech distortion.

Stated differently, a small forgetting factor results in a short memorywhich results in a faster convergence to the current noise condition.However, when speech activity starts, the noise coherence matrix islocked to the last computed value before the onset of speech. This valueis no longer a good representation of the noise condition during speechactivity due to the aggressive tracking that was used. On the otherhand, choosing a larger forgetting factor results in a slowerconvergence to the current noise. However, due to the large amount ofaveraging, when speech activity starts, the last computed noisecoherence matrix can perform well during the entirety of the speechsegment.

For example, a noise from the vehicle hitting a bump in the road or someother rapid transient noise will have a larger impact on a noisecoherence matrix with a smaller forgetting factor than on the noisecoherence with the larger forgetting factor. As a result, a noisecoherence matrix update with small forgetting factor will generally do abetter job adapting to the rapid transient than the noise coherencematrix update with a larger forgetting factor. However, if speech isbegun directly after the rapid transient noise, the noise coherencematrix will freeze for the duration of the speech having adapted to thetransient noise, which is no longer indicative of the noise conditionwithin the vehicle cabin. Thus, using the noise coherence matrix updatewith small forgetting factor works better for cancelling noise when theuser is not speaking, but the noise coherence matrix update with largeforgetting factor results in a more consistent results when the user isspeaking.

Accordingly, the benefits of having a small forgetting factor and alarge forgetting factor can be combined by computing two noise coherencematrices, one noise coherence matrix with a smaller forgetting factorand another with a larger forgetting factor, for each frequency bin, asshown in the following equations:

$\begin{matrix}{{C_{{nn},1}( {k,n} )} = \{ \begin{matrix}{{{{\lambda(k)} \cdot {C_{{nn},1}( {k,{n - 1}} )}} + \lbrack {1 - {\lambda(k)}} \rbrack},{\cdot {x( {k,n} )} \cdot {x^{H}( {k,n} )}},} & {{VAD} = 0} \\{\mspace{391mu}{{C_{{nn},1}( {k,{n - 1}} )},}} & {{VAD} = 1}\end{matrix} } & (9) \\{{C_{{nn},2}( {k,n} )} = \{ \begin{matrix}{{{{\lambda(k)} \cdot {C_{{nn},2}( {k,{n - 1}} )}} + {\lbrack {1 - {\lambda(k)}} \rbrack \cdot {x( {k,n} )} \cdot {x^{H}( {k,n} )}}},} & {{VAD} = 0} \\{\mspace{374mu}{{C_{{nn},2}( {k,{n - 1}} )},}} & {{VAD} = 1}\end{matrix} } & (10)\end{matrix}$

where λ₁(k) is a smaller forgetting factor and λ₂ (k) is a largerforgetting factor. It should be understood that, in this disclosure,“smaller” and “larger” forgetting factors are relative to each other.Thus, the smaller forgetting factor is smaller than the largerforgetting factor, and the larger forgetting factor is larger than thesmaller forgetting factors.

Equation (1) is modified to reflect the change as follows:

$\begin{matrix}{{w^{H}(k)} = \{ \begin{matrix}{\frac{{C_{{nn},1}^{- 1}(k)} \cdot {d(k)}}{{d^{H}(k)} \cdot {C_{{nn},1}^{- 1}(k)} \cdot {d(k)}},} & {{VAD} = 0} \\{\frac{{C_{{nn},2}^{- 1}(k)} \cdot {d(k)}}{{d^{H}(k)} \cdot {C_{{nn},2}^{- 1}(k)} \cdot {d(k)}},} & {{VAD} = 1}\end{matrix} } & (11)\end{matrix}$

Thus, in the absence of speech activity, C_(nn,1)(k) (calculated withthe smaller forgetting factor) is used to update the filter coefficientswhich results in a faster adaptation to the noise change and betterinterference cancellation. In the presence of speech activity,C_(nn,2)(k) (calculated with the larger forgetting factor) is used toupdate the filter coefficients which results in less speech degradationat the expense of a slightly lower interference cancellation.

This is shown in method 300 depicted in FIGS. 3A-3C, in which, at step304 and 306, noise coherence matrices with smaller and larger forgettingfactors are calculated. These can be computed according to Eqs. 9 and 10above, respectively. At step 308, it is determined whether voiceactivity is detected at the target seat. If not, then at step 310, theacoustic signal is estimated at the target seat, based on the microphonesignals m and the noise coherence matrix computed with the smallerforgetting factor. However, if voice activity is detected at the targetseat, then at step 310, the acoustic signal is estimated at the targetseat, based on the microphone signals and the noise coherence matrixcomputed with the larger forgetting factor.

As described above, the noise coherence matrix can be configured to beonly updated in the absence of speech activity. A robust speech activitydetector is required to prevent adaptation during speech segments.Unfortunately, it is impossible to identify the presence of speechbefore it occurs or even in the same sample in which it begins. Thespeech activity state will always be identified after its onset. Toaddress this issue, a history of noise coherence matrices computed atprevious frames can be stored in the memory. For example, let Δt be thetime the speech activity detector needs to identify speech activity.Once a speech activity state is detected, a previously stored noisecoherence matrix computed at least Δt before the identification ofspeech activity can be loaded from the memory (e.g., RAM) and used inthe computation of the filter coefficients while speech is ongoing.Generally speaking, only enough previously calculated noise coherencematrices need to be stored to permit retrieving a previously storednoise coherence matrix computed at some time Δt before the detection ofspeech.

This can be combined with the example described above, storing noisecoherence matrices with large and small forgetting factories by storinga history of noise coherence matrices with long forgetting factors. Oncethe user begins speaking, the noise coherence matrix with longerforgetting factor computed at least some length of time Δt before theidentification of speech activity can be loaded from the memory and usedin the computation of the filter coefficients.

This is depicted in step 314 of FIG. 3C, in which after voice activityis detected at the target seat in step 308, a noise coherence matrixcomputed with a larger forgetting factor, having been calculated at somepoint in time before the detection of the voice activity, is retrieved.This retrieved noise coherence matrix computed with larger forgettingfactor is used to estimate the acoustic signal at the target seat instep 312.

In addition to the noise coherence matrix computed with largerforgetting factor, the noise coherence matrix computed with smallerforgetting factor will be similarly corrupted by the user's voiceactivity before the voice activity detector has detected the user'sspeech. Thus, when speech activity ceases, a previously stored noisecoherence matrix computed with smaller forgetting factor can beretrieved from memory and used to estimate the acoustic signal at step310. The retrieved noise coherence matrix computed with smallerforgetting factor can be computed at some time before the identificationof speech activity. Like the above example, retrieving the noisecoherence matrix with smaller forgetting factor requires storing ahistory of noise coherence matrices computed with smaller forgettingfactors, such that one such noise coherence matrix computed at leastsome length of time Δt before the detection of the user's voice activitycan be retrieved. In most cases, this retrieved noise coherence matrixcomputed with smaller forgetting factor will be computed during the samesample that the retrieved noise coherence matrix with larger forgettingfactor (e.g., at step 314) is computed; however, in some examples, theretrieved noise coherence matrices can be computed during differentsamples. Once the noise coherence matrix computed with smallerforgetting factor is retrieved, it can either be used as the noisecoherence matrix (e.g., as C_(nn,1)(k) in Eq. (9)) or to initialize thenoise coherence matrix (e.g., as C_(nn,1)(k,n−1) in Eq. (9)) for thesample following the end of user speech.

FIG. 3D, which is situated in the NO branch of FIG. 3B, depicts thesteps of retrieving the noise coherence matrix with smaller forgettingfactor. At step 316 of FIG. 3D, if voice activity was detected at theprevious sample, then at step 318, a noise coherence matrix computedwith a smaller forgetting factor, having been calculated at some pointin time before the detection of the voice activity (beginning at orbefore the previous sample), is retrieved. This retrieved noisecoherence matrix computed with smaller forgetting factor is used toestimate the acoustic signal at the target seat in step 310. If, at step316, voice activity was not detected during the previous sample, then atstep 310, the noise coherence matrix with smaller forgetting factor,calculated according to the current sample and the previous sample, isused to estimate the acoustic signal at the target seating position.

While the methods for estimating the acoustic signal at the target seatusing smaller and larger forgetting factors is described for a singleseat, it should be understood that it can be extended to any number ofseats within the vehicle cabin.

The performance of the algorithm can suffer for a short period of timewhen a sudden change in the noise condition occurs. This can happen forexample when music is turned on or off, when a navigation prompt occurs,or when speech starts from an undesired location. During the convergencetime of the noise coherence to the new noise condition, the performanceof the array can deteriorate. To circumvent this, the principle ofsuperposition can be used to introduce adjustments to the noisecoherence matrix to better match the current noise condition and reducethe convergence time.

For example, at any given time, the overall noise-plus-interferencesignal can be expressed as follows:

$\begin{matrix}{{{n(t)} = {\sum\limits_{i = 1}^{N_{I}}\;{n_{i}(t)}}},} & (12)\end{matrix}$

where n_(i)(t) is the ith interference/noise signal and N_(I) is thetotal number of interference/noise signals. These signals can includeroad noise, wind noise, ventilation noise, navigation prompts, undesiredspeech, music, among others. Assuming that the interference signals areuncorrelated, i.e. <n_(i)(t), n_(j)(t)>=0 for i≠j, the overall noisecoherence matrix can be expressed as follows:

$\begin{matrix}{{{C_{nn}( {k,n} )} = {\sum\limits_{i = 1}^{N_{I}}\;{C_{{nn},i}( {k,n} )}}},} & (13)\end{matrix}$

where C_(nn,1)(k,n) is the noise coherence matrix of the ithinterference/noise signal.

This formulation can be used to update the noise coherence matrix when aknown interference signal is introduced to the vehicle cabin. In thisexample, noise coherence matrices associated with known signals, i.e.,music, navigation, etc., are predetermined (e.g., as a factory defaultset of known noise coherence matrices or as a set of noise coherencematrices that are calculated before the known signal is introduced tothe vehicle cabin) and added to the noise coherence matrix update asfollows:

C _(nn)(k,n ₀)=C _(nn)(k,n ₀−1)+g _(i) ·C _(nn,1)(k),  (14)

where n₀ denotes the frame at which the known interference signaln_(i)(t) appeared or disappeared, C_(nn,i)(k) is the noise coherencematrix associated with the interference signal, and g_(i) is a gain thatreflects the level of the interference signal. In other words, thepredetermined noise coherence matrix associated with a known signal issuperposed with the noise coherence matrix calculated from themicrophone signals m. For example, if voice navigation is queued to beplayed in the vehicle cabin, the noise coherence matrix can be updatedwith the predetermined noise coherence matrix during the same frame atwhich the navigation signal is played in the vehicle cabin. This permitsthe noise coherence matrix to be updated with the predetermined noisecoherence matrix of the known signal faster than the matrix coherencematrix could otherwise be updated from the microphone signals only. Oncethe noise coherence matrix is updated with the predetermined noisecoherence matrix, normal updating of the noise coherence matrix canresume.

This is depicted in the method 400 of FIG. 4A, in which, at step 404, anoise coherence matrix is computed based, in part, on a predeterminednoise coherence matrix that is representative of a known noise conditionintroduced to the vehicle cabin by at least one speaker in the vehiclecabin (e.g., navigation, music, etc.) As described above, thepredetermined noise coherence matrix can be a prestored factory defaultnoise coherence matrix or can be calculated while the known noisecondition is queued to be played in the vehicle cabin. The noisecoherence matrix can be computed, e.g., by superposing the calculatednoise coherence matrix with the predetermined noise coherence matrix.

Likewise, the predetermined noise coherence matrix can be subtractedfrom the calculated noise coherence matrix when the known is signal isno longer played in the vehicle cabin. This is particularly effectivewhen the known signal ceases to be played while the user is speaking,i.e., when the noise coherence matrix is not being updated. Otherwisefailing to subtract the noise coherence matrix of the known signal whenthe noise coherence matrix is not being updated results in an incorrectnoise coherence matrix that accounts for a signal not being played andthus does not perform properly.

Thus, at step 408, the noise coherence matrix is subtracted from theupdated noise coherence matrix. The updated noise coherence matrix isthe noise coherence matrix, as calculated from, at least, thepredetermined noise coherence matrix. This step occurs at a latersample, when the noise coherence matrix has been updated and the knownnoise condition ceases to be played in the vehicle cabin.

It should be noted that instead of adding or removing the correspondingnoise coherence matrix in a single frame, a short transition period overwhich the term g_(i)·C_(nn,i)(k) is added or removed can be used. Inother words, the gain g_(i) can be increased from a smaller value to alarger value over a plurality of frames to prevent sudden changes in thefilter coefficients. The final value of g_(i) can be determined by themagnitude of the known noise condition in the vehicle cabin 100.

In the case where the matrix inversion lemma is used to update theinverse of the noise coherence matrix (i.e., Eq. (3) instead of directmatrix inversion after Eq. (2)), the superposition method discussedabove in connection with Eq. (14) cannot be used to update the noisecoherence matrix.

Thus, in the proposed method, multiple solutions, where each solutionconsists of the inverses of the noise coherence matrices C_(nn) ⁻¹(k),are stored in the memory. Each solution corresponds to a set ofconditions (music level, HVAC level, vehicle speed, etc.). The storedsolutions can either be computed a priori or updated during run-time.When a known condition is encountered, the associated inverses can beloaded from the memory and the system then adapts the noise coherencematrix inverse using Eq. (3). In other words, rather than superposingthe calculated noise coherence matrix with a predetermined noisecoherence matrix, from which an inverse noise coherence matrix can thenbe calculated, the entire inverse noise coherence matrix (calculatedusing the matrix inversion lemma) is substituted for the predeterminedinverse noise coherence matrix when a known signal is to be introducedto the cabin.

This solution is, however, memory intensive because a multiplicity ofsolutions must be prestored, including for combinations of conditionssuch as soundstage setting, music level, HVAC level, vehicle speed,etc., as the combination of each of these conditions determines theinverse of the noise coherence matrix. In an alternative example, wherethe matrix inversion lemma is used, the value of the inverse of thenoise coherence matrix, calculated using the matrix inversion lemma, isinverted to yield the non-inverted noise coherence matrix. Thenon-inverted noise coherence matrix is then superposed with thepredetermined noise coherence matrix according to Eq. (14), at whichpoint this value is again inverted to result in the inverse of the noisecoherence matrix. Once the inverse of the superposed solution from Eq.(14) is determined, it can then be updated using the matrix inversionlemma according to new microphone signals m. This solution is morecomputationally intensive but requires fewer memory resources toimplement than substituting the inverted noise coherence matrix with apredetermined inverted noise coherence matrix.

Signal-to-noise ratio (SNR) improvement is one of the main metrics thatare used in evaluating the performance of a microphone array. Anotherperformance evaluation metric is the white noise gain (WNG). The whitenoise gain measures the ability of the microphone array to suppressnoncoherent noise. In other words, it is the SNR improvement provided bythe microphone array when the noise at the microphones is white noise.The SNR improvement and white noise gain are competing metrics.Designing a microphone array for optimal SNR improvement might result ina low (or even negative) white noise gain. This results in boosting thenoncoherent noise at the output of the array. On the other hand,improving the white noise gain of the array results in reducing theachievable SNR improvement.

Diagonal loading can be used to improve the white noise gain of an MVDRdesign. It can also improve the robustness of the design to changes inthe noise conditions. If the full matrix inversion is computed at everyframe, diagonal loading can be achieved by modifying Equation (1) asfollows:

$\begin{matrix}{{w^{H}(k)} = {\frac{\lbrack {{C_{nn}(k)} + {{\mu(k)} \cdot I}} \rbrack^{- 1} \cdot {d(k)}}{{d^{H}(k)} \cdot \lbrack {{C_{nn}(k)} + {{\mu(k)} \cdot I}} \rbrack^{- 1} \cdot {d(k)}}.}} & (15)\end{matrix}$

where μ(k) is a scalar value and I is an identity matrix. Diagonalloading adds a small value to the main diagonal of the noise coherencematrix. Increasing μ improves the white noise gain of the array at theexpense of lowering its SNR improvement. The diagonal loading value canbe constant across frequency bins or variable. Choosing a different μfor each frequency bin provides more flexibility in the design since thewhite noise gain of the array changes from one frequency bin to another.In the vehicle environment, lower frequencies tend to require largerdiagonal loading values than higher frequencies.

On the other hand, if the matrix inversion lemma is used to update theinverse of the noise coherence matrix, Equation (15) cannot be used toapply diagonal loading to the design. As a workaround, Eqs. (3) and (4)can be modified to achieve the same effect. Diagonal loading adds ascalar value to the main diagonal of the noise coherence matrix. This isequivalent to adding white noise with variance equal to the loadingvalue to the noise signal. Therefore, the observation vector x(k,n) ofthe microphone signals m can be modified as follows to add diagonalloading:

x _(DL)(k,n)=x(k,n)+√{square root over (μ(k))}·r _(WGN)(n)  (16)

where μ(k) is the desired diagonal loading value and r_(WGN)(n) is aN_(A)×1 vector whose elements are sampled from a Gaussian distributionwith mean zero and standard deviation one with N_(A) being the number ofmicrophones in the array. After enough averaging is done in Equations(3) and (4), the effective noise coherence matrix becomes equal toC_(nn)(k)+μ(k)·I. Stated differently a white noise signal (which isnoncoherent) can be added to each microphone signal (either at themicrophone signal or at the adaptive beamformer 112), such that thewhite noise signal is canceled.

This is shown in FIG. 5A, in which, at step 504 white noise isartificially added to each of the plurality of microphone signals (e.g.,according to Eq. (16)), such that the white noise gain is improved whenthe acoustic signal at the target seat is estimated at step 506. Asdescribed above, the white noise signal can be added to the microphonesignals or added at the adaptive beamformer 112.

In either the diagonal loading or added white noise examples, severalmethods can be used to set the values μ(k). For instance, the value ofμ(k) can be set to achieve a minimum white noise gain at each frequency.Alternatively, a condition number of the combined noise coherence matrixC_(nn)(k)+μ(k)·I, or of the noise coherence matrix after the white noisesignal is artificially added to the microphone signals 104, is used toset μ(k). The condition number is a metric that characterizes the degreeto which a given matrix can be inverted. Consequently, a conditionnumber that is too low will affect the performance of the array;whereas, a condition number that is too high will result in amplifyingnon-coherent noise for each frequency bin. Increasing μ(k) results inlowering the condition number of the combined noise coherence matrixwhich in turn results in a more robust MVDR design.

The value of μ(k) can be computed a priori and stored in the memory. Itcan consist of a single set of values {μ(k), k=1, . . . , N_(f)} whereN_(f) is the number of frequency bins. The set is designed to performwell with a variety of noise conditions. That is, in most noiseconditions, the condition number of the combined noise coherence or thenoise coherence matrix will be greater than a minimum condition numberand less than a maximum condition number. In other words, the same valueof μ(k) is used, per frequency bin, irrespective of the noise conditiondetected within the vehicle cabin. The minimum condition number is theminimum condition number allowable to maintain acceptable arrayperformance, such as, in one example, a condition number of 20. Themaximum condition number is a value selected according to maximumacceptable degree that the non-coherent noise can be amplified, such as,in one example, a condition number 100.

In an alternative example, multiple sets of diagonal loading values{μ(k), k=1, . . . , N_(f)} can also consist of multiple sets of valueswhere each set performs well in at least one noise condition. The setthat is used is picked based on the actual noise condition that isencountered in order to maintain the condition number between theminimum condition number and the maximum condition number, or tomaintain the condition number close to an ideal condition number (e.g.,70). Thus, the noise condition of detected microphone signals m dictatewhich set of values of μ(k) are used.

In an alternative example, the value of μ(k) can be updated in real-timeto adapt to the changes in the noise conditions. This can be done, forexample, by monitoring the condition number of the effective noisecoherence matrix, comparing it to a desired pre-determined conditionnumber, and adjusting the diagonal loading values accordingly.

Regarding the use of symbols herein, a capital letter, e.g., H,generally represents a term, signal, or quantity in the frequency orspectral domain, and a lowercase letter, e.g., h, generally represents aterm, signal, or quantity in the time domain. Relation between time andfrequency domain is generally well known, and is described at leastunder the realm of Fourier mathematics or analysis, and is accordinglynot presented herein. Additionally, signals, transfer functions, orother terms or quantities represented by symbols herein may be operated,considered, or analyzed in analog or discrete form. In the case of timedomain terms or quantities, the analog time index, e.g., t, and/ordiscrete sample index, e.g., n, may be interchanged or omitted invarious cases. Likewise, in the frequency domain, analog frequencyindexes, e.g, f, and discrete frequency indexes, e.g., k, are omitted inmost cases. Further, relationships and calculations disclosed herein maygenerally exist or be carried out in either time or frequency domains,and either analog or discrete domains, as will be understood by one ofskill in the art. Accordingly, various examples to illustrate everypossible variation in time or frequency domains, and analog or discretedomains, are not presented herein.

The functionality described herein, or portions thereof, and its variousmodifications (hereinafter “the functions”) can be implemented, at leastin part, via computer program product, e.g., a computer program tangiblyembodied in an information carrier, such as one or more non-transitorymachine-readable media or storage device, for execution by, or tocontrol the operation of, one or more data processing apparatus, e.g., aprogrammable processor, a computer, multiple computers, and/orprogrammable logic components.

A computer program can be written in any form of programming language,including compiled or interpreted languages, and it can be deployed inany form, including as a stand-alone program or as a module, component,subroutine, or other unit suitable for use in a computing environment. Acomputer program can be deployed to be executed on one computer or onmultiple computers at one site or distributed across multiple sites andinterconnected by a network.

Actions associated with implementing all or part of the functions can beperformed by one or more programmable processors executing one or morecomputer programs to perform the functions of the calibration process.All or part of the functions can be implemented as, special purposelogic circuitry, e.g., an FPGA and/or an ASIC (application-specificintegrated circuit).

Processors suitable for the execution of a computer program include, byway of example, both general and special purpose microprocessors, andany one or more processors of any kind of digital computer. Generally, aprocessor will receive instructions and data from a read-only memory ora random access memory or both. Components of a computer include aprocessor for executing instructions and one or more memory devices forstoring instructions and data.

While several inventive embodiments have been described and illustratedherein, those of ordinary skill in the art will readily envision avariety of other means and/or structures for performing the functionand/or obtaining the results and/or one or more of the advantagesdescribed herein, and each of such variations and/or modifications isdeemed to be within the scope of the inventive embodiments describedherein. More generally, those skilled in the art will readily appreciatethat all parameters, dimensions, materials, and configurations describedherein are meant to be exemplary and that the actual parameters,dimensions, materials, and/or configurations will depend upon thespecific application or applications for which the inventive teachingsis/are used. Those skilled in the art will recognize, or be able toascertain using no more than routine experimentation, many equivalentsto the specific inventive embodiments described herein. It is,therefore, to be understood that the foregoing embodiments are presentedby way of example only and that, within the scope of the appended claimsand equivalents thereto, inventive embodiments may be practicedotherwise than as specifically described and claimed. Inventiveembodiments of the present disclosure are directed to each individualfeature, system, article, material, and/or method described herein. Inaddition, any combination of two or more such features, systems,articles, materials, and/or methods, if such features, systems,articles, materials, and/or methods are not mutually inconsistent, isincluded within the inventive scope of the present disclosure.

1. An adaptive beamforming system, comprising: a plurality ofmicrophones disposed about a vehicle cabin, each of the plurality ofmicrophones generating a microphone signal, wherein the vehicle cabindefines a plurality of seating positions; a voice-activity detectorconfigured to detect when, at least, a user seated in a target seat ofthe plurality of seating positions is speaking; and an adaptivebeamformer configured to receive the microphone signals from theplurality of microphones and to generate, based on the microphonesignals and a noise coherence matrix, an estimate of an acoustic signalat the target seating position, according to an adaptive beamformingalgorithm, wherein the noise coherence matrix ceases to be updated when,according to the voice-activity detector, the user seated in the targetseating position is speaking.
 2. The system of claim 1, wherein thetarget seat is selected according to a user selection.
 3. The system ofclaim 1, wherein the target seat is selected according to which seatingposition of the plurality of seating positions the voice activitydetector detects a user speaking.
 4. The system of claim 1, wherein theadaptive beamformer is further configured to cease updating the noisecoherence matrix when, according to the voice-activity detector, a userin any of the plurality of seating positions is speaking.
 5. The systemof claim 1, wherein the adaptive beamformer is further configured tocalculate a plurality of noise coherence matrices, each of the pluralityof noise coherence matrices being representative of a noise condition ata respective associated seating position of the plurality of seatingpositions, wherein each of the plurality of noise coherence matricesceases to be updated when, according to the voice-activity detector, auser seated in the associated seating position is speaking.
 6. Thesystem of claim 1, wherein the adaptive beamformer is further configuredto generate an estimate of a second acoustic signal at a second targetseating position, when, according to the voice-activity detector, theuser seated in the target seating position is speaking and the userseated in the second target seating position is speaking, wherein theestimate of the acoustic signal and the second acoustic signal beingsummed together.
 7. The system of claim 1, wherein the adaptivebeamformer is further configured to generate an estimate of the acousticsignal and a second acoustic signal at a second target seating position,according to a second adaptive beamforming algorithm, when, according tothe voice-activity detector, the user seated in the target seatingposition is speaking and the user seating in the second target seatingposition is speaking.
 8. The system of claim 7, wherein the secondadaptive beamforming algorithm is a linearly constrained minimumvariance beamforming algorithm.
 9. An adaptive beamforming system,comprising: a plurality of microphones disposed about a vehicle cabin,each of the plurality of microphones generating a microphone signal,wherein the vehicle cabin defines a plurality of seating positions; avoice-activity detector configured to detect when a user seated in atarget seating position of the plurality of seating positions isspeaking; and an adaptive beamformer configured to receive themicrophone signals from the plurality of microphones and to generate,based on the microphone signals and one of a first noise coherencematrix and a second noise coherence matrix, an estimate of an acousticsignal at the target seating position, according to an adaptivebeamforming algorithm, wherein the first noise coherence matrix iscomputed by recursively summing a first previously-calculated noisecoherence matrix with a first newly-calculated noise coherence matrix,wherein the first previously-calculated noise coherence matrix isweighted more heavily than the first newly-calculated noise coherencematrix, wherein coefficients of the adaptive beamformer are updatedusing the first noise coherence matrix when the voice-activity detectordetects that the user seated in the target seating position is speaking,wherein the second noise coherence matrix is computed by recursivelysumming a second previously-calculated noise coherence matrix with asecond newly-calculated noise coherence matrix, wherein the secondpreviously-calculated noise coherence matrix is weighted less heavilythan the second newly-calculated noise coherence matrix, wherein thecoefficients of the adaptive beamforming filter are updated using thesecond noise coherence matrix when the voice-activity detector does notdetect that the user seated in the target seating position is speaking.10. The system of claim 9, wherein the coefficients of the adaptivebeamformer are updated using a historic first noise coherence matrixstored during a previous frame, wherein the historic first noisecoherence matrix was stored at least a predetermined period of timebefore the voice-activity detector detected that the user seated in thetarget seating position is speaking, wherein the predetermined period oftime is greater than a delay required for the voice-activity detector todetect that the target user is speaking.
 11. An adaptive beamformingsystem, comprising: a plurality of microphones disposed about a vehiclecabin, each of the plurality of microphones generating a microphonesignal, wherein the vehicle cabin defines a plurality of seatingpositions; and an adaptive beamformer configured to receive themicrophone signals from the plurality of microphones and to generate,based on the microphone signals and a noise coherence matrix, anestimate of an acoustic signal at a target seating position, accordingto an adaptive beamforming algorithm, wherein the noise coherence matrixis determined, at least in part, from a predetermined noise coherencematrix, the predetermined noise coherence matrix being representative ofa noise condition introduced to the vehicle cabin by at least onespeaker disposed in the vehicle cabin.
 13. The system of claim 11,wherein a gain of the predetermined noise coherence matrix is setaccording to a magnitude of the noise condition.
 14. The system of claim13, wherein the gain of the predetermined noise coherence matrix isincreased over a plurality of samples.
 15. The system of claim 11,wherein the predetermined noise coherence matrix is subtracted from anupdated noise coherence matrix when the noise condition ceases to beintroduced to the vehicle cabin, wherein the updated noise coherencematrix is an update to the noise coherence matrix according to theplurality of microphone signals.
 16. The system of claim 11, wherein thenoise coherence matrix comprises a sum of a calculated noise coherencematrix and the predetermined noise coherence matrix, wherein thecalculated noise coherence matrix is inverted from an inverted statebefore being summed with the predetermined noise coherence matrix. 17.The system of claim 11, wherein the predetermined noise coherence matrixis retrieved as an inverted matrix.
 18. An adaptive beamforming system,comprising: a plurality of microphones disposed about a vehicle cabin,each of the plurality of microphones generating a microphone signal,wherein the vehicle cabin defines a plurality of seating positions; anadaptive beamformer configured to receive the microphone signals fromthe plurality of microphones and to generate, based on the microphonesignals and a noise coherence matrix, an estimate of an acoustic signalat a target seating position, according to an adaptive beamformingalgorithm, wherein an artificial white noise signal is added to each ofthe plurality of microphone signals such that white noise gain of theestimate of the acoustic signal is improved.
 19. The system of claim 18,wherein the artificial white noise signal is selected such that aminimum white noise gain is achieved across a predetermined frequencyrange.
 20. The system of claim 18, wherein the artificial white noisesignal is predetermined.
 21. The system of claim 18, wherein theartificial white noise signal is selected from a plurality of artificialwhite noise signals according to the noise condition within the cabin.22. The system of claim 18, wherein the artificial white noise signal isadjusted such that a condition number of the noise coherence matrix ismaintained within a predetermined range across frequency.